http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html
http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpsession.html
will help you
also you can use the README file of gstrtpbin.
Think that will be good to start...after that you can read RFC's
All the best....Please share your findings as well as querries, that is how
most of us learns :)
Thanks a lot
Manish
On Sun, Aug 17, 2008 at 8:21 PM, Tiago Katcipis <[EMAIL PROTECTED]>wrote:
> hmm thanks for the help guys, my last question is where to start? i know
> that i can always try google, but maybe i will loose time reading the wrong
> stuff, if you point me on the right direction i would appreciate.
>
> best regards
>
> tiago
>
>
> On Sun, Aug 17, 2008 at 7:02 AM, Manish Rana <[EMAIL PROTECTED]>wrote:
>
>> hey generally it depends on the network dealys............
>> but i think generally 1000ms is an Ok ok but RTCP will genreally let u
>> know the jitter value
>>
>> On Sat, Aug 16, 2008 at 7:23 PM, Tiago Katcipis <[EMAIL PROTECTED]>wrote:
>>
>>> and how bigger would have to be the jitter buffer? i cant use on the lib
>>> a too big jitter.
>>>
>>> thanks everyone for the help
>>>
>>>
>>> On Sat, Aug 16, 2008 at 5:25 AM, Manish Rana <[EMAIL PROTECTED]>wrote:
>>>
>>>> hey but i setting the jitter-buffer latency we can take care of the
>>>> round trip delays
>>>> and alsa will get mostly sequenced audio packets....
>>>>
>>>>
>>>>
>>>> On Sat, Aug 16, 2008 at 9:04 AM, gulshan karmani <
>>>> [EMAIL PROTECTED]> wrote:
>>>>
>>>>> hi all,
>>>>> Only issue cud be voice rendering which has constraints of round trip
>>>>> delays due to naec and in that case use of alsa plugin cud be a
>>>>> problem.
>>>>> Rgds,
>>>>> Gulshan
>>>>>
>>>>> On 8/15/08, Manish Rana <[EMAIL PROTECTED]> wrote:
>>>>> > Hi,
>>>>> >
>>>>> > as far as the real time constrain is concerned, this should be taken
>>>>> care by
>>>>> > RTP and add the required delays shall be added by RTP, that is
>>>>> gstrtpbin
>>>>> > plugin in gstreamer.
>>>>> > On addition to this gstreamer will give u flexiblity to create the
>>>>> pipelines
>>>>> > as your requirements. You can have minimal elements as well or add
>>>>> more to
>>>>> > get the better audio. (like audio resample and audioconvert can be
>>>>> optional)
>>>>> >
>>>>> > And if i am not wrong Gstreamer is used successfully in maemo for the
>>>>> VoIP
>>>>> > application, and there is Farsight plugin available, which is
>>>>> optimised.
>>>>> >
>>>>> > I am sorry if I have any wrong info... Please correct me....
>>>>> >
>>>>> > Also please add more on the same...........
>>>>> >
>>>>> > BR
>>>>> > Manish
>>>>> > On Fri, Aug 15, 2008 at 3:58 PM, Tiago Katcipis <
>>>>> [EMAIL PROTECTED]>wrote:
>>>>> >
>>>>> >> I'm working in a project using voip on a software and in embedded
>>>>> systems,
>>>>> >> its more like a lib, that is used by a high level software for PC
>>>>> and in
>>>>> >> embedded systems. Actually everything is done in one single gigantic
>>>>> >> function, now we are working on creating a more readable and
>>>>> expandable
>>>>> >> lib,
>>>>> >> so we started to build the lib using pipes and filters patterns.
>>>>> That's
>>>>> >> when
>>>>> >> the idea of using gstreamer came, but since gstreamer is usually
>>>>> used on
>>>>> >> media players i would like to know if it is good to be used on real
>>>>> time
>>>>> >> voip systems that rely heavily on time to work properly. Is
>>>>> Gstreamer a
>>>>> >> good
>>>>> >> lib to build this type of application? If it is who would be the
>>>>> best
>>>>> >> place
>>>>> >> for me to start reading about it (using gstreamer on voip) ?
>>>>> >>
>>>>> >> sorry if i asked something stupid... I'm just starting on the job
>>>>> and
>>>>> >> don't
>>>>> >> have to much experience, sorry for the lousy English too :-)
>>>>> >>
>>>>> >> Best regards
>>>>> >>
>>>>> >> Tiago César Katcipis
>>>>> >>
>>>>> >>
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>>>>>
>>>>
>>>>
>>>
>>
>
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